[Asterisk-Users] Junghans Quad-BRI card and asterisk cvs-head
We're using the Quad-BRI card from Junghanns.NET with corresponding drivers (bristuff 0.0.2). The driver tries to patch asterisk libpri, which fails for current version. Anyone got an idea what'S the latest version of asterisk / libtri usable with the Quad-BRI Card? Thanks, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 Call Transfers
We're using a couple of h323 IP Phones (innovaphone ip200) w/ asterisk. Basic call setup works, but we can't get call transfers to work: On pressing the transfer button on the phone (getting a new dialtone) the 2nd endpoint is disconnected. Any idea if we can get this to work? Same reaction using the innovaphone ip400 gatekeeper and using gnugk. Asterisk version is 0.7.2 release. Thanks, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Junghans Quad-BRI card and asterisk cvs-head
The bristuff distribution comes with a install.sh script (./install.sh) which downloads, compiles the required software on your system. If you want to do it manually, look into download.sh to see the exact cvs checkout options which downloads the required asterisk and libpri versions. Yes, I know which libpri/asterisk versions bristuff downloads when using the included scripts (03/24/04). Problem is, I'd like to get the features / bugfixes from later versions. I'd especially like to try current oh323 drives, which require cvs head and don't compile against the versions usd by bristuff 0.2.2. Is it possible to combine older libtri with cvs-head asterisk or is that just asking for trouble? Thanks, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] H323 Call Transfers
How about running a current (cvs -head) version of Asterisk? Would love to and of course tried to: no go because of Junghans Quad-BRI ISDN Card, no driver for cvs -head. Bye, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: AW: [Asterisk-Users] Junghans Quad-BRI card and asterisk cvs-head
If you can't wait you can use the patch from someone who merged the bristuff patch with a more recent version of cvs head... This one: http://capi4linux.thepenguin.de/download/asterisk/bri-stuff-0. 0.2a-pp.tar.gz Thanks for that pointer, I'll give it a try. Bye, martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] * and Innovaphone
Hi Torsten, I think I have the same problem as Martin Bene mentioned in http://lists.digium.com/pipermail/asterisk-users/2004-January/ 034521.html Since I found no further information about this I'd like to ask wether you know what the reason for this problem is and how one can get around this. I've since spent some time debugging the problem: The innovaphone gatekeeper hands out a bandwidth allocation of 8kbit on registration; I haven't found any way to deactivate or configure this limit. Two possible workarounds: * Don't have asterisk register any extensions with the gatekeeper * Or, as an utterly ugly workaround, I've hacked the openh323 libs to ignore the bandwidth limit and proceed andway. Seems to work OK. Bye, Martin openh323_bandwidth.patch Description: openh323_bandwidth.patch
AW: [Asterisk-Users] IP Phones that support G.723 on H.323
Does anyone know of Phone that supports G.723 on H.323. Innovaphone tiptel 200 for example. http://www.innovaphone.com/webneu2/products/en_IP200.asp One of the nicest phones I've seen so far, h.323 only though. Bye, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Cisco 7960 SIP Images
I had a completely different experience. The day I decided I wanted to get a contract, I called Cisco, gave them my personal credit card, and three hours later had my CCO access upgraded. I just bought a smartnet for one phone for two years (a whopping $16), there was nothing to it. Nope, same exerience as Johns here. Runaround trying to find a reseller, got a softnet (not smartnet) contract for ~ EUR8 that allows access to CCO; no information available on what else this might include. Waited two weeks for the Service Tokens to arrive by mail only to find out that the online registration site listed on the contracts doesn't know how to handle softnet. Waited another 2 1/2 Weeks for cisco to manually activate the contracts. Cisoc data sheet for the 7940 states: Other Cisco IP Phone 7940G features include: G.711 and G.729a audio compression H.323 compatible and Microsoft NetMeeting compatibility When asked about h.323 compatibility git told that the phone works with cisco call manager, and since that supports h.323, it enables the phone to work with h.323... No comment necessary I think. Bye, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] I got it (was: Cisco 7940 with asterisk)
Hi Siggi/Jan, If so, there's still a load version conflict (although I've never seen a 7960 or 7940 care about the version communicated through SCCP): On the phone, press Settings, then 4 for load information. watch out for the App-Load-ID. On my 7940, this is P00305000300. Yours is most likely a smaller number... If you have a CVS version of chan_sccp (either Jan's CVS or Theo's), just add this exact string to your /etc/asterisk/sccp.conf in the device definition for your phone, ie. something like: I've had the same problem (looping resets) with my 7940. The Error Verifying Config Info Message doesn't have anything to do with the real problem. I also get that message, possibly because I don't keep a device specific config file (SEP000D65707B78.cnf.xml) or DISTINCTIVERINGLIST.XML on my tftp server. The real problem is keepalive timing: * the 7940 doesn't like the 5 second default timeout - set timeout to =10 seconds and you avoid the restart loop. Bye, Martin Martin Bene, CTO icomedias GmbH, A-8020 Graz, Entenplatz 1b t +43 (316) 721671-14, f +43 (316) 721671-26 e [EMAIL PROTECTED], i http://www.icomedias.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: h323 with innovaphone ip 400 gatekeeper/innovaphone Ip200 phones
I know - talking to myself is a bad sign, but still: While asterisk can receive audio from the h323 phone quite fine, reverse doesn't work. Added information: when turning up h.323 tracing, I get this: channels.cxx(777) LogChan Bandwidth requested/used = 64.0/0.0 kb/s h323.cxx(4419)H323Bandwidth request: -0.0kb/s, available: 32.0kb/s h323.cxx(4419)H323Bandwidth request: +64.0kb/s, available: 32.0kb/s h323.cxx(4425)H323Available bandwidth exceeded The 64kb/s is the bandwidth requirement for g711u (which is the selected codec); I don't understand where the available bandwidth of 32kb/sec comes from though, but it's definitely the cause for my failures. Now to find out where that limit comes from.. Bye, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Again: 7920 Cisco IP Phone Skinny SIP
Hi Jan, in the sccp_registration i would then handle the registration for the 7920 how the callmanager is behaving. I've just gotten one step further with my 7920: Got it successfully registered to asterisk. Still doesn't actually work, but definitely a step in the right direction. The problem is that the 7920 expects SelectSoftKeys Messages to finish its setup - probably it sends the offhook/onhook sequence to trigger these. Asterisk however doesn't send these until it's finished registering the device on reception of a message of type 2d - which the 7920 never sends. Hack/workaround, against current cvs: diff -urN chan_sccp/sccp_actions.c chan_sccp.mbe/sccp_actions.c --- chan_sccp/sccp_actions.c2004-01-17 11:18:36.0 +0100 +++ chan_sccp.mbe/sccp_actions.c2004-01-17 17:14:15.0 +0100 @@ -94,7 +94,7 @@ } } - sccp_dev_set_registered(d, RsProgress); + sccp_dev_set_registered(d, RsOK); d-currentLine = d-lines; REQ(r1, RegisterAckMessage); This probably breaks all kind of other things, but it does let the 7920 register with asterisk. Next Problem: Softkeys need special attention, currently the onhook/offhook keys aren't mapped correctly, so you can't actually accept a call or hang up :-) Dialing works bzw. Bye, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Re: Again: 7920 Cisco IP Phone Skinny SIP
Hi, [...] There should also be a digitally signed version of that file (cmterm_7920.*.sbn), which the phone probably requires. nope. no sbn. according to my cisco source the file is not signed. Funny, that would be the first phone with unsigned firmware. But I'll double-check after the next firmware update. At least for my other phones, Cisco introduced signed binaries for versions = 5.0; looks like the 7920 firmware is still below that. Bye, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] More Success on the Cisco 7920 and SCCP !!!!!
Hi Siggi, 7960 and then Call Ended on the Display (curious about that !!!). That seems to be normal for the 7920. I've sniffed the registration procedure with Cisco's newest 3.3(3) CallManager (+patches), and it's doing the same thing. Maybe that's some odd way of testing if the CallManager (CCM) really works... After that the phone reboots and the stuff repeats Same thing here. CCM does quite a few things in different order compared to chan_sccp, but apart from that, the registration procedure seems quite similar. I'm still looking into the detailed differences (which is a bit hard, as there doesn't seem to be any tool like diff for ethereal traces). Since I've got a 7920 myself and am trying to get things to work: If you've still got access to the cisco stuff: could you make available a tcpdump file (tcpdump -w) of a successfull callmanager registration? I'd really like to see what the successfull tftp and skinny sessions look like and try to duplicate that w/ asterisk. Thanks, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Problems with Cisco 7920/Skinny/Asterisk
Hi Jan, the last 2 days i was working on getting the 7920 Phones to work with Skinny Asterisk; however no luck (yet). Same here, no joy so far. Does anybody has a SEPDefault.CNF.xml and a SEPmac.CNF.xml handy for me ? it should be documented at the cisco page, but it isn't :-( The default file should only get requested when the specific file wasn't found. The ony slightly relevant thing I've found so far is a sample file in the chan_sccp package (http://www.zozo.org.uk/pages.shtml?page=sccp), a 2nd implementation for skinny protocol (and no, I can't get that to work either). I still have the issue that the 7920 spits out No Service - IP Config failed but Asterisk is giving me sign that the phone has registered. Slightly differnt to what happens here - I get No Callmanager found and the phone goes back to IP configuration, also after having registered at the call manager. RECEIVED UNKNOWN MESSAGE TYPE: 2b This isn't your problem, that's just a headset status message. so what is buggy ? Any hints/ideas ? What's really needed at this point would be a tcpdump -w or ethereal trace of the connection + tftp sequence between 7920 and a real cisco callmanager - anyone got access to such a configuration? I should get access to a test environment, but it'll take some weeks. Bye, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
Hi Richard, Load balancers have some added value, but those that have had to deal with a problem where a single system within the cluster is up but not processing data would probably argue their actual value. I've done quite a lot of work with clustered/ha linux configurations. I usualy try to keep additional boxes/hardware to an absolute minimum, otherwise the newly introduced points of (hardware) failure tend to make the whole exersize pointless. A solution I found to work quite well: Software load balancer (using LVS) run as a HA service (ldirectord) on two of the servers. This allows use of quite specific probes for the real servers being balanced, so a server not correctly processing requests can be removed from the list of active quite reliably. Since the director script is perl, adding probes for protocols not supported in the default install is fairly streightforward. If any proposed design actually involved a different MAC address, obviously all local sip phones would die since the arp cache timeout within the phones would preclude a failover. (Not cool.) Arp cache timeouts usualy don't come into this: when moving a cluster IP address to a different NIC (probaly on a different machine) you can broadcast gratuitous arp packets on the affected ethernet segment; this updates the arp caches of all connected devices and allows failovers far faster than arp chache timeout. Notable exception: some firewalls can be quite paranoid wrt. to arp updates and will NOT accept gratuitous arp packets. I've run into this with a cluster installation with one of my customers. Technology now supports 100 meg layer-2 pipes throughout a city at a reasonable cost. If a cluster were split across mutiple buildings within a city, it certainly would be of interest to those that are responsible for business continuity planning. Are there limitations? I'm wary of split cluster configurations because often the need for multiple, independent communication paths between cluster nodes gets overlooked or ignored in these configurations, greatly increasing risk of split-brain configurations, i.e. several nodes in the cluster thinking they're the only online server and trying to take over services. This easily/usually leads to a real mess (data corruption) that can be costly to clean up. When keeping your nodes in physical proximity it's much easier to have, say, 2 network links + one serial link between cluster nodes thus providing a very resilient fabric for inter-cluster communications. Someone mentioned the only data needed to be shared between clustered systems was phone Registration info (and then quickly jumped to engineering a solution for that). Is that the only data needed or might someone need a ton of other stuff? (Is cdr, iax, dialplans, agi, vm, and/or other dynamic data an issue that needs to be considered in a reasonable high-availability design?) Depends on what you want/need to fail over in case your asterisk box goes down. in stages that'd be 1 (cluster) IP address for sip/h323 etc. services 2 voice mail, recordings, activity logs 3 registrations for connected VoIP clients 4 active calls (VoIP + PSTN) For the moment, item 4 definitely isn't feasible; even if we get some hardware to switch over E1/T1/PRI whatever interfaves, card or interface initialisation will kill active calls. Item 2 would be plain file on-disk data; for an active/standby cluster replicating these should be pretty straigthforward using either shared storage or an apropriate filesystem/blockdevice replication system. I've personaly had good experience with drbd (block device replication over the network; only supports 2 nodes in active/standby configuration but works quite well for that.) Item 3 should also feasible; this information is already persistent over asterisk restarts and seems to be just a berkley db file for a default install. Sme method as for item 2 should work. I'd have to guess there are probably hundreds on this list that can engineer raid drives, ups's for ethernet closet switches, protected cat 5 cabling, and switch boxes that can move physical interfaces between servers. But, I'd also guess there are far fewer that can identify many of the sip, rtp, iax, nat, cdr, etc, etc, issues. What are some of those issues? (Maybe there aren't any?) Since I'm still very much an asterisk beginner I'll have to pass on this one; However, I'm definitely going to do some experiments on my test cluster systems with asterisk to just see what breaks when failing over asterisk services. Also, things get MUCH more interesting when yo start to move from plain active/standby to active/active configurations: here, for failover, you'll end up with the registration and file data from the failed server and need to integrate that into an already running server merging the seperate sets of information - preferably without trashing the running server :-) Bye, Martin
[Asterisk-Users] asterisk sccp support
Hi zozo, sorry to bother you; I've been trying to get a cisco 7920 phone (that's the cute wifi/wireless modell) to work with asterisk (no success so far) and ran across your chan_sccp module; is there a current cvs version that I could base my tests on? tar version 0.1 seems to be ~2003-09-12, cvs access as posted on http://theo.me.uk/pages.shtml?page=sccp doesn't seem to work any more: Logging in to :pserver:[EMAIL PROTECTED]:2401/var/lib/cvs CVS password: /var/lib/cvs: no such repository Nevertheless: I started with the 0.1 driver. first problem: The 7920 Phone doesn't want to play if VersionReqMessage isn't implemented; OK, so I added that (P00603010033; I'm not sure if the final digits for the firmware are actually right). Better result now: I get as far as Registered with Asterisk PBX on the phone, only to have it continue to no Callmanager found and retry (ad infinitum). Here's what I get from asterisk debug output: == Got message RegisterMessage Auto logging into cisco1 == Sending Packet Type DisplayPromptStatusMessage (48 bytes) == Sending Packet Type RegisterAckMessage (24 bytes) == Sending Packet Type CapabilitiesReqMessage (4 bytes) == Got message IpPortMessage == Got message HeadsetStatusMessage == Got message VersionReqMessage == Sending Packet Type VersionMessage (20 bytes) == Got message CapabilitiesResMessage Device has 6 Capabilities -- CODEC: 4 - G.711 u-law 64k -- CODEC: 2 - G.711 A-law 64k -- CODEC: 11 - G.729 -- CODEC: 12 - G.729 Annex A -- CODEC: 15 - G.729 Annex B -- CODEC: 16 - G.729 Annex A+Annex B == Got message ButtonTemplateReqMessage WARNING[262161]: File sccp_actions.c, Line 144 (sccp_handle_button_template_req): Don't have a button layout, sending blank template. == Sending Packet Type ButtonTemplateMessage (100 bytes) == Got message SoftKeyTemplateReqMessage == Sending Packet Type SoftKeyTemplateResMessage (656 bytes) == Got message SoftKeySetReqMessage -- Set[0] = 0:11 -- -- Set[1] = 0:1 1:2 2:10 -- -- Set[2] = 0:1 1:6 2:12 -- -- Set[3] = 0:12 1:13 2:14 3:6 -- -- Set[4] = 0:-1 1:6 -- -- Set[5] = 0:3 1:6 2:7 3:8 4:-1 5:9 -- -- Set[6] = 0:3 1:4 -- -- There are 7 SoftKeySets. == Sending Packet Type SoftKeySetResMessage (784 bytes) == Sending Packet Type KeepAliveAckMessage (4 bytes) == Sending Packet Type KeepAliveAckMessage (4 bytes) == Sending Packet Type KeepAliveAckMessage (4 bytes) == Sending Packet Type KeepAliveAckMessage (4 bytes) WARNING[262161]: File sccp_socket.c, Line 42 (sccp_read_data): No length in read: Success Looks like the phone goes south on receipt of the SoftKeySetResMessage; any idea of how to continue from here? Thanks for your time for reading so far, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
